[alsa-devel] Notebook Toshiba satellite x200-21g with realtek snd-hda-intel audio device still not supported (codec ALC268)
hi all i'm new to this mailing list i've bought a laptop and i would like to help you supporting my sound card
what works: 2 speaker, low volume (it should be a 5+1 audio card)
what doesn't -) low volume -) inoperating headphone jack -) not working mic -) no subwoofer -) may be some other stuff not tested ( SPDIF? hardware midi?)
with
speaker-test -t wav -c 6
only front left/right channel works
my kernel is a debian kernel Linux Luffy 2.6.23-1-686-bigmem #1 SMP Wed Dec 5 02:42:05 UTC 2007 i686 GNU/Linux
alsa 1.0.15 drivers (compiled with debian-way: module-assistant a-i alsa-source)
i've already tried all the option i could think about
auto (default) toshiba 3stacks 3stacks-6ch 3stacks-6ch-dig
neither one has better/worst behavior
I've checked the module has toke those option by cat /sys/module/snd_hda_intel/parameters/model
anyway I've read the ALSA-Configuration.txt shipped with kernel and for Codec ALC268 it only has 4 options ALC268 3stack 3-stack model toshiba Toshiba A205 acer Acer laptops auto auto-config reading BIOS (default)
I've googled a bit and i found out that the Toshiba A205 has completely different problem.. the sound doesn't work with it.. reading the changelog from 1.0.14 to 1.0.15 (alsa) it say: Added the proper model=toshiba for Toshiba A305 with ALC268 codec.
so i think my audio card is an unknown model the laptop is brand new and i only found this model in Italy anyway i think that the same audio card is shipped with the Toshiba Satellite x205-S7483 witch is an american laptop
some info about the audio card:
lspci -n
00:1b.0 0403: 8086:284b (rev 03)
lspci -vvv
00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Toshiba America Info Systems Unknown device ff00 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at f6600000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5)
cat /proc/asound/card0/codec#0
Codec: Realtek ALC268 Address: 0 Vendor Id: 0x10ec0268 Subsystem Id: 0x1179ff0a Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x02 Node 0x0f [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x1d Node 0x10 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] Connection: 3 0x03 0x1d 0x02 Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0f Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x0221101f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP Connection: 1 0x10 Node 0x16 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x02a11830: [Jack] Mic at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x24: IN Connection: 1 0x02 Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: Pincap 0x083724: IN Detect Pin Default 0x99a30931: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x24: IN Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x0281103e: [Jack] Line In at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Connection: 1 0x02 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x4016852d: [N/A] Speaker at Ext N/A Conn = Digital, Color = Purple Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x0810: OUT Pin Default 0x02451120: [Jack] SPDIF Out at Ext Front Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x23 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x1f 0x1f] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x12 Node 0x24 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x8a 0x8a] Connection: 7 0x18 0x19 0x1a* 0x1c 0x14 0x15 0x13
many of this think doesn't mean a thing to me but i will be glad to help you if you ask.. i don't mind compiling some other driver/kernel/tool for this just ask me
one last note: searching the internet i've found this page: https://wiki.ubuntu.com/Gutsy_Intel_HD_Audio_Controller
if you go down you find that the Toshiba Satellite x205-S7483 has the exact same problem here. I think they have the same card
searching the Realtek web site i didn't find any 6 channel audio card with the ALC268 codec may be this mean the ALC268 is not the right codec for my audio card or this is a really new model/heavily Toshiba modified device
i've already filed a bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3659
please ask me if you need any more info or want me to do some test
thanks you all for your work
ps: i've subscribed to vger mailing list, what's the right mailing list for alsa-devel?
Daniele (Mastro) ha scritto:
i've already filed a bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3659
please ask me if you need any more info or want me to do some test
i just want to appoint that this email is a bit different from the one i've written at the vger mailing list (this is for Takashi Iwai who replied on vger and may be didn't read this new message)
now i'm using alsa 1.0.15 with the toshiba model parameter and i still have the same problems
and again: i've no idea on what you could need.. i really want to help you supporting my card: it's useful to me in the first place after all
so ask me if you need something
and sorry if my english isn't perfect! I'm not english and i know i still have some problem with it
thank you all for your time
-- Daniele Segato
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1
Daniele (Mastro) ha scritto:
and again: i've no idea on what you could need.. i really want to help you supporting my card: it's useful to me in the first place after all
so ask me if you need something
and sorry if my english isn't perfect! I'm not english and i know i still have some problem with it
thank you all for your time
i don't want to be rude, i only want to ask you if is there something wrong with my emails? nobody is replying anymore. do you plan to reply?
some information missing? i'll provide it, just tell me what you need
if you plan to reply but haven't had the time yet don't be upset for this message... i don't know how things works here and i don't want to put you in a hurry, i just want to know if you will do something for this problem or not?
thanks again
- -- Daniele
At Thu, 20 Dec 2007 09:46:04 +0100, Daniele (Mastro) wrote:
Daniele (Mastro) ha scritto:
and again: i've no idea on what you could need.. i really want to help you supporting my card: it's useful to me in the first place after all
so ask me if you need something
and sorry if my english isn't perfect! I'm not english and i know i still have some problem with it
thank you all for your time
i don't want to be rude, i only want to ask you if is there something wrong with my emails? nobody is replying anymore. do you plan to reply?
some information missing? i'll provide it, just tell me what you need
More time and hands to work :)
if you plan to reply but haven't had the time yet don't be upset for this message... i don't know how things works here and i don't want to put you in a hurry, i just want to know if you will do something for this problem or not?
I recommend you to hack the driver code by yourself. The coding itself isn't too difficult. The problem is the guess work, which PIN really corresponds to which I/O, and what control is missing (EAPD, GPIO, etc).
What I'm wondering is how to support 5.1 output. The ALC268 codec is at most for 4 channels (2 + 2). Are you sure that you can get 5.1 distinct outputs?
Takashi
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1
Takashi Iwai ha scritto:
I recommend you to hack the driver code by yourself.
i take this as a "no we will not do it"
but i understand it.. keep reading
The coding itself isn't too difficult. The problem is the guess work, which PIN really corresponds to which I/O, and what control is missing (EAPD, GPIO, etc).
ok... i know a bit of C programming, not much to be honest.. but i have no idea on where/how to watch for "PIN" and things like that... I don't know what EAPD, GPIO etc. are i really would like to do it myself but i don't know where to start... if you can give me a link that explain how it work and help me in the process i'll do my best
but without any help i don't know what to do.. and i'm worried to damage my audio card if i do something wrong...
What I'm wondering is how to support 5.1 output. The ALC268 codec is at most for 4 channels (2 + 2). Are you sure that you can get 5.1 distinct outputs?
hum.. you put a doubt in me.. i'm sure there's a subwoofer but i'm not sure it should be 6 channel... i try to check in some way.. i never booted up Vista in this notebook... i guess i could try it to check this or may be i found something googling... i let you know..
i'm a bit sad that you haven't time/men for my chipset (specially for the headphone jack not working and the mic) but i understand it... i hope you will help me to learn and fix my problem alone helping the comunity and your project
i'll wait your reply best regards :) - -- Daniele
Daniele,
Normally I would have jumped in earlier to help, but I am swamped, between my real job (non-alsa related), school (BS IT degree program), and recent flooding activity in Oregon a couple of weeks ago.
What I can do, is ask you to download and run http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh .
Also, rebuild the modules with "./configure --with-cards=snd-hda-intel --with-debug=detect" and load them with "model=auto". This will leave output in dmesg as to what the driver finds in bios. It's not always reliable, but it helps.
As to 5.1 surround, the only way to get 6 channel output would be through the SPDIF port, as it acts as a digial passthrough. If your system has a subwoofer, it is very likely tied into the main speaker out, just like the 2.1 speakers you can get at most stores.
Tobin
On Fri, 2007-12-21 at 01:04 +0100, Daniele (Mastro) wrote:
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1
Takashi Iwai ha scritto:
I recommend you to hack the driver code by yourself.
i take this as a "no we will not do it"
but i understand it.. keep reading
The coding itself isn't too difficult. The problem is the guess work, which PIN really corresponds to which I/O, and what control is missing (EAPD, GPIO, etc).
ok... i know a bit of C programming, not much to be honest.. but i have no idea on where/how to watch for "PIN" and things like that... I don't know what EAPD, GPIO etc. are i really would like to do it myself but i don't know where to start... if you can give me a link that explain how it work and help me in the process i'll do my best
but without any help i don't know what to do.. and i'm worried to damage my audio card if i do something wrong...
What I'm wondering is how to support 5.1 output. The ALC268 codec is at most for 4 channels (2 + 2). Are you sure that you can get 5.1 distinct outputs?
hum.. you put a doubt in me.. i'm sure there's a subwoofer but i'm not sure it should be 6 channel... i try to check in some way.. i never booted up Vista in this notebook... i guess i could try it to check this or may be i found something googling... i let you know..
i'm a bit sad that you haven't time/men for my chipset (specially for the headphone jack not working and the mic) but i understand it... i hope you will help me to learn and fix my problem alone helping the comunity and your project
i'll wait your reply best regards :)
Daniele -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
iD8DBQFHawMcgSF24OYDe4YRAgSsAKCw/UD/OIfZIN1uz54EYFaxR9VC3QCg466b nDJ0hV9iaSWmAt3LzeqtrhM= =ldxH -----END PGP SIGNATURE----- _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1
Tobin Davis ha scritto:
Daniele,
Normally I would have jumped in earlier to help, but I am swamped, between my real job (non-alsa related), school (BS IT degree program), and recent flooding activity in Oregon a couple of weeks ago.
i'm swamped too for many reason and haven't much time... anyway i don't have hurry.. i just would like my audio card to work one day :)
What I can do, is ask you to download and run http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh .
Also, rebuild the modules with "./configure --with-cards=snd-hda-intel --with-debug=detect" and load them with "model=auto". This will leave output in dmesg as to what the driver finds in bios. It's not always reliable, but it helps.
i'll do this and keep you informed, thanks for the advice (i forgot to mention my card has a build in MIDI support... should i compile it with the option sequencer? and why don't you put inside the "oss" too?
As to 5.1 surround, the only way to get 6 channel output would be through the SPDIF port, as it acts as a digial passthrough. If your system has a subwoofer, it is very likely tied into the main speaker out, just like the 2.1 speakers you can get at most stores.
i still didn't search for this.. i'll do! i can tell you i've 4 speaker on the front of my notebook (may be they are only 2 channel) and a subwoofer on the bottom of it
i'll bring here as much info i can find on my audio system and will start to try what you said... but i'll do this in spare time so may be it will take some time
thank you again!
- -- Daniele
Tobin Davis ha scritto:
What I can do, is ask you to download and run http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh .
that's the one with model=toshiba (useless) http://pastebin.ca/831128
Also, rebuild the modules with "./configure --with-cards=snd-hda-intel --with-debug=detect" and load them with "model=auto". This will leave output in dmesg as to what the driver finds in bios. It's not always reliable, but it helps.
done.. i recompiled all the alsa-source with that option (--with-debug=detect)
you can see dmesg here: http://pastebin.ca/831221 and the result of the alsa-info.sh script with this modules/setting here: http://pastebin.ca/831224
i've added 2 note to the bug ( https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3659 )
still i don't know how to use the info from dmesg for fixing my card... can you help me understanding it?
As to 5.1 surround, the only way to get 6 channel output would be through the SPDIF port, as it acts as a digial passthrough. If your system has a subwoofer, it is very likely tied into the main speaker out, just like the 2.1 speakers you can get at most stores.
as you can read into the bug report i think it's NOT a 5+1 sound card there are 2 middle frequency speaker, 2 twitter and 1 subwoofer (a total of 5 speaker) but probably, as you said, it's a 2.1 or something like this...
anyway it doesn't work as it should
i can recompile the driver with -with-debug=full option if you need it
i didn't compiled only the hda-intel driver.. i recompiled all using debian way (module assistant + modification to the debian/rules script)
merry Christmas and thanks for your time
Takashi Iwai ha scritto:
I recommend you to hack the driver code by yourself. The coding itself isn't too difficult. The problem is the guess work, which PIN really corresponds to which I/O, and what control is missing (EAPD, GPIO, etc).
hum.. i guess the file i should edit is patch_realtek.c but i really have no idea on what to change/add and how to understand what i have to change...
in my other message there are the debug info you requested and i don't know how to use them
i'm really want to do something to help you but i keep reading the code without knowing what to do or understanding it very much... i never developed kernel drivers and I've no idea on what to do
I'm putting here my good will but think i miss the knowledge to do what you suggested and i don't know where to achieve it either
What I'm wondering is how to support 5.1 output. The ALC268 codec is at most for 4 channels (2 + 2). Are you sure that you can get 5.1 distinct outputs?
probably this isn't a 5.1 as i said in the other message
thanks for any reply you will give me i'm sorry i'm so annoying
I haven't forgotten about you (yet). :)
I'll look at getting you a patch this week. Work is scheduled to be very slow due to the holidays.
Tobin
On Wed, 2007-12-26 at 17:08 +0100, Daniele (Mastro) wrote:
Takashi Iwai ha scritto:
I recommend you to hack the driver code by yourself. The coding itself isn't too difficult. The problem is the guess work, which PIN really corresponds to which I/O, and what control is missing (EAPD, GPIO, etc).
hum.. i guess the file i should edit is patch_realtek.c but i really have no idea on what to change/add and how to understand what i have to change...
in my other message there are the debug info you requested and i don't know how to use them
i'm really want to do something to help you but i keep reading the code without knowing what to do or understanding it very much... i never developed kernel drivers and I've no idea on what to do
I'm putting here my good will but think i miss the knowledge to do what you suggested and i don't know where to achieve it either
What I'm wondering is how to support 5.1 output. The ALC268 codec is at most for 4 channels (2 + 2). Are you sure that you can get 5.1 distinct outputs?
probably this isn't a 5.1 as i said in the other message
thanks for any reply you will give me i'm sorry i'm so annoying
Ok, I'm going to dive into this one this weekend. I have a test board ready to run.
Daniele, please email me off list to keep the noise level down. I'm also tracking this in bugzilla.
Tobin
On Wed, 2007-12-26 at 17:08 +0100, Daniele (Mastro) wrote:
Takashi Iwai ha scritto:
I recommend you to hack the driver code by yourself. The coding itself isn't too difficult. The problem is the guess work, which PIN really corresponds to which I/O, and what control is missing (EAPD, GPIO, etc).
hum.. i guess the file i should edit is patch_realtek.c but i really have no idea on what to change/add and how to understand what i have to change...
in my other message there are the debug info you requested and i don't know how to use them
i'm really want to do something to help you but i keep reading the code without knowing what to do or understanding it very much... i never developed kernel drivers and I've no idea on what to do
I'm putting here my good will but think i miss the knowledge to do what you suggested and i don't know where to achieve it either
What I'm wondering is how to support 5.1 output. The ALC268 codec is at most for 4 channels (2 + 2). Are you sure that you can get 5.1 distinct outputs?
probably this isn't a 5.1 as i said in the other message
thanks for any reply you will give me i'm sorry i'm so annoying
Tobin Davis ha scritto:
Ok, I'm going to dive into this one this weekend. I have a test board ready to run.
Daniele, please email me off list to keep the noise level down. I'm also tracking this in bugzilla.
since 16/01 i haven't any news from Tobin Davis in private email..
i sent him 3 mails in 3 weeks waiting for a reply without gettin any one
you can find all the news at the bug report:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3659
the last time i sent him an email with some remind on what's are, still, my problems..
can someone else take care of this problem? i'll try to help as best as i can... you can find on the bug report the debug ( detect ) info with "test" module parameter
note: things still missing.. microphone don't work in recording, the 56k modem isn't recognized and i've some error in dmesg: ALSA /usr/src/modules/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1784: chipset global capabilities = 0x4401 ALSA /usr/src/modules/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:736: codec_mask = 0xb ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:2212: hda_codec: model 'auto' is selected ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:2855: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:2859: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:2863: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:2871: inputs: mic=0x19, fmic=0x18, line=0x1a, fline=0x0, cd=0x0, aux=0x0 ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:1075: Cannot find slave Surround Playback Volume, skipped ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:1075: Cannot find slave Center Playback Volume, skipped ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:1075: Cannot find slave LFE Playback Volume, skipped ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:1075: Cannot find slave Side Playback Volume, skipped ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:1075: Cannot find slave Speaker Playback Volume, skipped ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:1075: Cannot find slave Mono Playback Volume, skipped ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:1075: Cannot find slave iSpeaker Playback Volume, skipped ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:1075: Cannot find slave Line-Out Playback Volume, skipped ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:1075: Cannot find slave Surround Playback Switch, skipped ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:1075: Cannot find slave Center Playback Switch, skipped ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:1075: Cannot find slave LFE Playback Switch, skipped ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:1075: Cannot find slave Side Playback Switch, skipped ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:1075: Cannot find slave Speaker Playback Switch, skipped ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:1075: Cannot find slave Mono Playback Switch, skipped ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:1075: Cannot find slave iSpeaker Playback Switch, skipped ALSA /usr/src/modules/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1264: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:691: hda_codec_setup_stream: NID=0x6, stream=0x5, channel=0, format=0x11 ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:691: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 ALSA /usr/src/modules/alsa-driver/pci/hda/hda_codec.c:691: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11
thanks again
At Mon, 11 Feb 2008 16:46:55 +0100, Daniele (Mastro) wrote:
Tobin Davis ha scritto:
Ok, I'm going to dive into this one this weekend. I have a test board ready to run.
Daniele, please email me off list to keep the noise level down. I'm also tracking this in bugzilla.
since 16/01 i haven't any news from Tobin Davis in private email..
i sent him 3 mails in 3 weeks waiting for a reply without gettin any one
you can find all the news at the bug report:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3659
the last time i sent him an email with some remind on what's are, still, my problems..
can someone else take care of this problem? i'll try to help as best as i can... you can find on the bug report the debug ( detect ) info with "test" module parameter
note: things still missing.. microphone don't work in recording, the 56k modem isn't recognized and i've some error in dmesg:
Note that these are no errors but just debug information.
Try the patch below. This should fix the capture source problem on ALC268.
The modem is a different problem. There are only certain model chips that can work with slmodemd. Non si3054-compatible (typically conexant ones) don't work.
Takashi
---
diff -r 29ae582df0ee pci/hda/patch_realtek.c --- a/pci/hda/patch_realtek.c Mon Feb 11 16:35:02 2008 +0100 +++ b/pci/hda/patch_realtek.c Mon Feb 11 17:06:19 2008 +0100 @@ -238,6 +238,7 @@ struct alc_spec { /* capture */ unsigned int num_adc_nids; hda_nid_t *adc_nids; + hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */
/* capture source */ @@ -291,6 +292,7 @@ struct alc_config_preset { hda_nid_t hp_nid; /* optional */ unsigned int num_adc_nids; hda_nid_t *adc_nids; + hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; unsigned int num_channel_mode; const struct hda_channel_mode *channel_mode; @@ -337,9 +339,10 @@ static int alc_mux_enum_put(struct snd_k struct alc_spec *spec = codec->spec; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); unsigned int mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx; + hda_nid_t capsrc = spec->capsrc_nids ? + spec->capsrc_nids[adc_idx] : spec->adc_nids[adc_idx]; return snd_hda_input_mux_put(codec, &spec->input_mux[mux_idx], ucontrol, - spec->adc_nids[adc_idx], - &spec->cur_mux[adc_idx]); + capsrc, &spec->cur_mux[adc_idx]); }
@@ -708,6 +711,7 @@ static void setup_preset(struct alc_spec
spec->num_adc_nids = preset->num_adc_nids; spec->adc_nids = preset->adc_nids; + spec->capsrc_nids = preset->capsrc_nids; spec->dig_in_nid = preset->dig_in_nid;
spec->unsol_event = preset->unsol_event; @@ -9527,6 +9531,10 @@ static hda_nid_t alc268_adc_nids_alt[1] 0x08 };
+static hda_nid_t alc268_capsrc_nids[2] = { + 0x23, 0x24 +}; + static struct snd_kcontrol_new alc268_base_mixer[] = { /* output mixer control */ HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), @@ -10140,6 +10148,7 @@ static struct alc_config_preset alc268_p .dac_nids = alc268_dac_nids, .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), .adc_nids = alc268_adc_nids_alt, + .capsrc_nids = alc268_capsrc_nids, .hp_nid = 0x03, .dig_out_nid = ALC268_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc268_modes), @@ -10154,6 +10163,7 @@ static struct alc_config_preset alc268_p .dac_nids = alc268_dac_nids, .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), .adc_nids = alc268_adc_nids_alt, + .capsrc_nids = alc268_capsrc_nids, .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc268_modes), .channel_mode = alc268_modes, @@ -10169,6 +10179,7 @@ static struct alc_config_preset alc268_p .dac_nids = alc268_dac_nids, .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), .adc_nids = alc268_adc_nids_alt, + .capsrc_nids = alc268_capsrc_nids, .hp_nid = 0x02, .num_channel_mode = ARRAY_SIZE(alc268_modes), .channel_mode = alc268_modes, @@ -10197,6 +10208,7 @@ static struct alc_config_preset alc268_p .dac_nids = alc268_dac_nids, .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), .adc_nids = alc268_adc_nids_alt, + .capsrc_nids = alc268_capsrc_nids, .hp_nid = 0x03, .dig_out_nid = ALC268_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc268_modes), @@ -10214,6 +10226,7 @@ static struct alc_config_preset alc268_p .dac_nids = alc268_dac_nids, .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), .adc_nids = alc268_adc_nids_alt, + .capsrc_nids = alc268_capsrc_nids, .hp_nid = 0x03, .dig_out_nid = ALC268_DIGOUT_NID, .num_channel_mode = ARRAY_SIZE(alc268_modes), @@ -10289,6 +10302,7 @@ static int patch_alc268(struct hda_codec alc268_capture_mixer; spec->num_mixers++; } + spec->capsrc_nids = alc268_capsrc_nids; }
spec->vmaster_nid = 0x02;
At Mon, 11 Feb 2008 17:09:14 +0100, I wrote:
At Mon, 11 Feb 2008 16:46:55 +0100, Daniele (Mastro) wrote:
Tobin Davis ha scritto:
Ok, I'm going to dive into this one this weekend. I have a test board ready to run.
Daniele, please email me off list to keep the noise level down. I'm also tracking this in bugzilla.
since 16/01 i haven't any news from Tobin Davis in private email..
i sent him 3 mails in 3 weeks waiting for a reply without gettin any one
you can find all the news at the bug report:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3659
the last time i sent him an email with some remind on what's are, still, my problems..
can someone else take care of this problem? i'll try to help as best as i can... you can find on the bug report the debug ( detect ) info with "test" module parameter
note: things still missing.. microphone don't work in recording, the 56k modem isn't recognized and i've some error in dmesg:
Note that these are no errors but just debug information.
Try the patch below. This should fix the capture source problem on ALC268.
Err, forget it. This won't fix anything.
Upload the output of alsa-info.sh to here. Make sure that you adjusted the mixer setting properly before doing this, of course.
thanks,
Takashi
Takashi Iwai ha scritto:
Note that these are no errors but just debug information.
ok.. so, are they useful?
Try the patch below. This should fix the capture source problem on ALC268.
ok i'll try as soon as i can.. with what "model" param should i create the alsa-info.sh log? test? default?
The modem is a different problem. There are only certain model chips that can work with slmodemd. Non si3054-compatible (typically conexant ones) don't work.
i think i was not very clear about this... i can't see anywhere something like a modem on lspci or lsusb... i think the modem is "inside" the audio card and for some reason it is ignored... this is why i think it's alsa-related
have i explained myself now?
At Mon, 11 Feb 2008 23:58:33 +0100, Daniele (Mastro) wrote:
Takashi Iwai ha scritto:
Note that these are no errors but just debug information.
ok.. so, are they useful?
Not really in your case.
Try the patch below. This should fix the capture source problem on ALC268.
ok i'll try as soon as i can.. with what "model" param should i create the alsa-info.sh log? test? default?
Basically run alsa-info.sh on your problematic configuration you wanted to use. So, don't pass model option unless you did so in the earlier versions and it ran well. If you have another better-working configuration, take alsa-info.sh snapshot as well for comparison.
The modem is a different problem. There are only certain model chips that can work with slmodemd. Non si3054-compatible (typically conexant ones) don't work.
i think i was not very clear about this... i can't see anywhere something like a modem on lspci or lsusb... i think the modem is "inside" the audio card and for some reason it is ignored... this is why i think it's alsa-related
Sure, and as I wrote, the modem is a *different* problem from the audio problem you are facing. I'd be intereseted if it's a regression. But in other cases, it'll be likely ignored, as not fixable properly right now.
Takashi
Takashi Iwai ha scritto:
Basically run alsa-info.sh on your problematic configuration you wanted to use. So, don't pass model option unless you did so in the earlier versions and it ran well. If you have another better-working configuration, take alsa-info.sh snapshot as well for comparison.
this is with "default" configuration... and it's the best one... has this the things you need?
Sure, and as I wrote, the modem is a *different* problem from the audio problem you are facing. I'd be intereseted if it's a regression. But in other cases, it'll be likely ignored, as not fixable properly right now.
i see... well i hope it will be supported one day thanks anyway i wish i could write them myself
Daniele (Mastro) ha scritto:
Takashi Iwai ha scritto:
Basically run alsa-info.sh on your problematic configuration you wanted to use. So, don't pass model option unless you did so in the earlier versions and it ran well. If you have another better-working configuration, take alsa-info.sh snapshot as well for comparison.
this is with "default" configuration... and it's the best one... has this the things you need?
with "default" configuration i mean model=auto and.. i forgot to raise capture volume:
if i raise the second capture volume (capture 1) i got playback sound (even if i disable it) and record is a noisy sound... if you want i can upload it somewhere
i've retoken the alsa-info http://pastebin.ca/905273
i've noticed that now on aplay -L i have
default:CARD=Intel HDA Intel, ALC268 Analog Default Audio Device front:CARD=Intel,DEV=0 HDA Intel, ALC268 Analog Front speakers surround40:CARD=Intel,DEV=0 HDA Intel, ALC268 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Intel,DEV=0 HDA Intel, ALC268 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Intel,DEV=0 HDA Intel, ALC268 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Intel,DEV=0 HDA Intel, ALC268 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Intel,DEV=0 HDA Intel, ALC268 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers null Discard all samples (playback) or generate zero samples (capture)
but if i try to test surround with: speaker-test -c6 -Dplug:surround51 -t wav -l1
i get:
speaker-test 1.0.15
Playback device is plug:surround51 Stream parameters are 48000Hz, S16_LE, 6 channels WAV file(s) Playback open error: -16,Device or resource busy Playback open error: -16,Device or resource busy Playback open error: -16,Device or resource busy [....]
i don't know how to test the digital output/input
and i've no "mix, mux or aux" on capture device list (only mic, front mic and line)
Daniele (Mastro) ha scritto:
Daniele (Mastro) ha scritto:
Takashi Iwai ha scritto:
Basically run alsa-info.sh on your problematic configuration you wanted to use. So, don't pass model option unless you did so in the earlier versions and it ran well. If you have another better-working configuration, take alsa-info.sh snapshot as well for comparison.
if i raise the second capture volume (capture 1) i got playback sound (even if i disable it) and record is a noisy sound... if you want i can upload it somewhere
i've retoken the alsa-info http://pastebin.ca/905273
is the alsa-info.sh missing something?
or is there any other problem?
At Mon, 18 Feb 2008 19:06:01 +0100, Daniele (Mastro) wrote:
Daniele (Mastro) ha scritto:
Daniele (Mastro) ha scritto:
Takashi Iwai ha scritto:
Basically run alsa-info.sh on your problematic configuration you wanted to use. So, don't pass model option unless you did so in the earlier versions and it ran well. If you have another better-working configuration, take alsa-info.sh snapshot as well for comparison.
if i raise the second capture volume (capture 1) i got playback sound (even if i disable it) and record is a noisy sound... if you want i can upload it somewhere
i've retoken the alsa-info http://pastebin.ca/905273
is the alsa-info.sh missing something?
or is there any other problem?
Just lack of time. I checked and fixed the problem right now on HG tree. Give it a try. Take HG repo from http://hg.alsa-project.org/alsa-kernel
(hg-mirror might be out of sync, so better to use the above one.)
Takashi
Takashi Iwai ha scritto:
Just lack of time.
no problem.. i was only worried that this got ignored again or that i didn't provided the right info
I checked and fixed the problem right now on HG tree. Give it a try. Take HG repo from http://hg.alsa-project.org/alsa-kernel
(hg-mirror might be out of sync, so better to use the above one.)
can you tell me what you watched in the alsa-info log to fix it? i'll try the new version as soon as i can!
At Tue, 19 Feb 2008 15:18:49 +0100, Daniele (Mastro) wrote:
Takashi Iwai ha scritto:
Just lack of time.
no problem.. i was only worried that this got ignored again or that i didn't provided the right info
I checked and fixed the problem right now on HG tree. Give it a try. Take HG repo from http://hg.alsa-project.org/alsa-kernel
(hg-mirror might be out of sync, so better to use the above one.)
can you tell me what you watched in the alsa-info log to fix it? i'll try the new version as soon as i can!
The fix is basically the widget connection of NID 0x23 (and 0x24). This was uninitialized and thus pointed the wrong widget. Unfortunately, this connection couldn't be changed because you apparently have only one input source. The patch fixes this issue.
Takashi
Takashi Iwai ha scritto:
The fix is basically the widget connection of NID 0x23 (and 0x24). This was uninitialized and thus pointed the wrong widget. Unfortunately, this connection couldn't be changed because you apparently have only one input source. The patch fixes this issue.
tried the new patch (i've compiled a today snapshot and then manually downloaded the last patch_realtek.c version)
the "front mic" capture now works.. i've 2 control: front mic and mic.. only mic works but i've a lot of crappy noise... http://www.mediafire.com/?c1c1jly9n7e
download this little .wav file i recorded with:
arecord -vv -f S16_LE -c1 -r48000 -d 5 test.wav
NOTE: while recording i see the "bar" always at 99%
thanks for your support.. do you have any idea on what's the cause of that noise?
At Tue, 19 Feb 2008 16:03:18 +0100, Daniele (Mastro) wrote:
Takashi Iwai ha scritto:
The fix is basically the widget connection of NID 0x23 (and 0x24). This was uninitialized and thus pointed the wrong widget. Unfortunately, this connection couldn't be changed because you apparently have only one input source. The patch fixes this issue.
tried the new patch (i've compiled a today snapshot and then manually downloaded the last patch_realtek.c version)
the "front mic" capture now works.. i've 2 control: front mic and mic.. only mic works but i've a lot of crappy noise... http://www.mediafire.com/?c1c1jly9n7e
download this little .wav file i recorded with:
arecord -vv -f S16_LE -c1 -r48000 -d 5 test.wav
NOTE: while recording i see the "bar" always at 99%
thanks for your support.. do you have any idea on what's the cause of that noise?
The mixer setting? You should set "Digital Capture Volume" 50%, corresponding to 0dB. The capture volume doesn't have to be too high, too.
The vref level could be a different one depending on the mic device you are using. But VREF80 is OK in most cases.
Takashi
Takashi Iwai ha scritto:
The mixer setting? You should set "Digital Capture Volume" 50%, corresponding to 0dB. The capture volume doesn't have to be too high, too.
oh.. i had to play a bit with alsamixer but finally i got a good result...
my setting: - front mic boost = 50 (option: 0, 50, 100) - Capture enabled = 65 - Capture 1 disabled (see below) - Digital = 68 - Input Source = Front mic - Input Source 1 = Front mic (playback - see below)
i've discovered some oddity:
- i have 2 capture device.... (capture and capure 1) - the first one is *really* a capture device and work only if i select "front mic", not "mic" - the second one is the mic playback device... and again it work only if i use "front mic"
should it work this way?
The vref level could be a different one depending on the mic device you are using. But VREF80 is OK in most cases.
i don't know what "VREF80" but now capturing with mic work...
now remaining problems: - midi hardware (my card should have it): i've loaded snd-seq-midi module and i've an alsa hardware device but i can't play midi files (they play but i hear nothing...). i've created another alsa-info log taken with the snd-seq-midi loaded http://pastebin.ca/909849 - no "mux","mix", or "aux" record source to directly record from desktop - surround still doesn't work i should have surround $ aplay -L default:CARD=Intel HDA Intel, ALC268 Analog Default Audio Device front:CARD=Intel,DEV=0 HDA Intel, ALC268 Analog Front speakers surround40:CARD=Intel,DEV=0 HDA Intel, ALC268 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Intel,DEV=0 HDA Intel, ALC268 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Intel,DEV=0 HDA Intel, ALC268 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Intel,DEV=0 HDA Intel, ALC268 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Intel,DEV=0 HDA Intel, ALC268 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers null Discard all samples (playback) or generate zero samples (capture)
but trying speaker-test -c5 -Dplug:surround41
or any of that other devices produce this error: Playback device is plug:surround41 Stream parameters are 48000Hz, S16_LE, 5 channels Using 16 octaves of pink noise Playback open error: -16,Device or resource busy
the last error repeat itself until i stop it (CTRL+C) and i hear nothing
are some of this fixable? and can i help in some way fixing them? thanks again... A LOT!
At Tue, 19 Feb 2008 16:55:02 +0100, Daniele (Mastro) wrote:
Takashi Iwai ha scritto:
The mixer setting? You should set "Digital Capture Volume" 50%, corresponding to 0dB. The capture volume doesn't have to be too high, too.
oh.. i had to play a bit with alsamixer but finally i got a good result...
my setting:
- front mic boost = 50 (option: 0, 50, 100)
- Capture enabled = 65
- Capture 1 disabled (see below)
- Digital = 68
Keep this 50%. It's a digital attenuation/gain control and changing this makes no sense unless you have a digital mic without the hardware volume control.
- Input Source = Front mic
- Input Source 1 = Front mic (playback - see below)
Is "Front mic" the external mic jack? And the internal mic works as well?
i've discovered some oddity:
- i have 2 capture device.... (capture and capure 1)
- the first one is *really* a capture device and work only if i select
"front mic", not "mic"
- the second one is the mic playback device... and again it work only if
i use "front mic"
should it work this way?
The second capture could work independently, assigned to PCM#2.
The vref level could be a different one depending on the mic device you are using. But VREF80 is OK in most cases.
i don't know what "VREF80" but now capturing with mic work...
now remaining problems:
- midi hardware (my card should have it): i've loaded snd-seq-midi
module and i've an alsa hardware device but i can't play midi files (they play but i hear nothing...). i've created another alsa-info log taken with the snd-seq-midi loaded http://pastebin.ca/909849
No, you are fooled by marketing people. It has no MIDI hardware. It's a pure software thingy.
- no "mux","mix", or "aux" record source to directly record from desktop
The codec has no function to record from the mixer amp (some other Realtek codecs have, though). It's again a software issue.
- surround still doesn't work
i should have surround
No, you don't. aplay just shows the available "configuration" but it the device doesn't support surrounds.
Takashi
Takashi Iwai ha scritto:
- Digital = 68
Keep this 50%. It's a digital attenuation/gain control and changing this makes no sense unless you have a digital mic without the hardware volume control.
if i put 50% there the mic it's a bit "low" with volume and i have to spoke really near to it (however sound is cleaner, lower but cleaner)
- Input Source = Front mic
- Input Source 1 = Front mic (playback - see below)
Is "Front mic" the external mic jack? And the internal mic works as well?
"front mic" works as the jack in the front of my laptop no internal mic doesn't work at all.. and i don't know if there's one to be honest cause i never booted windows here
The second capture could work independently, assigned to PCM#2.
sorry i think i didn't understand what do you mean.. so the behavior is "normal"? or it isn't? or it is "fixable" in some way? with my previous laptop i used to have only 1 capture device and to select if i want to use it only as playback, only as capture or both..
here i've to do this separately with 2 device.. it's not a big deal.. it's only strange
No, you are fooled by marketing people. It has no MIDI hardware. It's a pure software thingy.
actually it say under "audio specification" for my notebook
Standard sonori supportati : MIDI support
which means: Supported sound standard: MIDI support
you can read it here: http://it.computers.toshiba-europe.com/cgi-bin/ToshibaCSG/jsp/SUPPORTSECTION... (i don't know if this page will still visible in future)
- no "mux","mix", or "aux" record source to directly record from desktop
The codec has no function to record from the mixer amp (some other Realtek codecs have, though). It's again a software issue.
ok.. so i've no way to "record what I'm listening" right? (no way other that using a microphone off course)
No, you don't. aplay just shows the available "configuration" but it the device doesn't support surrounds.
oh ok :) do you know how can i test subwoofer? it seems active but if there's a way to actually "test" it would be nice
sorry for me bothering you :) i hope one day I'll be able to pay back your kindness -- Daniele
The only way to support Midi on that system is via software midi. There is no hardware midi support directly to the system (no place to plug in midi devices), except through USB or possibly integrated in a docking system. To get Midi support for Linux, use Timidity (http://timidity.sourceforge.net). This will convert midi sequences to pcm audio tracks.
I personally can't translate the specs listed, but from what I can tell, the system should have a Line-in, Mic-In, and HP-out jacks, along with 4 internal speakers (nothing indicating surround sound). It does have SPDIF out, and I would assume that will allow surround sound if hooked up to a digital receiver, as SPDIF is essentially a digital pass-through. The codec itself only has two DACs, which could be configured between the headphones and speakers to do a quasi-surround4.0 setup, but based on the chip diagram would take some additional programming effort.
And I'm sorry I dropped the ball in helping here. I wish I could spend more time on Alsa, but economics forces me to have a full time job doing non-alsa work at this time. Bad excuse, I know.
I'll try to get my test board working again this week, and I'll even download the Windows drivers for your system to use as a baseline for comparison. No guarantees, though.
Tobin
On Tue, 2008-02-19 at 18:29 +0100, Daniele (Mastro) wrote:
Takashi Iwai ha scritto:
- Digital = 68
Keep this 50%. It's a digital attenuation/gain control and changing this makes no sense unless you have a digital mic without the hardware volume control.
if i put 50% there the mic it's a bit "low" with volume and i have to spoke really near to it (however sound is cleaner, lower but cleaner)
- Input Source = Front mic
- Input Source 1 = Front mic (playback - see below)
Is "Front mic" the external mic jack? And the internal mic works as well?
"front mic" works as the jack in the front of my laptop no internal mic doesn't work at all.. and i don't know if there's one to be honest cause i never booted windows here
The second capture could work independently, assigned to PCM#2.
sorry i think i didn't understand what do you mean.. so the behavior is "normal"? or it isn't? or it is "fixable" in some way? with my previous laptop i used to have only 1 capture device and to select if i want to use it only as playback, only as capture or both..
here i've to do this separately with 2 device.. it's not a big deal.. it's only strange
No, you are fooled by marketing people. It has no MIDI hardware. It's a pure software thingy.
actually it say under "audio specification" for my notebook
Standard sonori supportati : MIDI support
which means: Supported sound standard: MIDI support
you can read it here: http://it.computers.toshiba-europe.com/cgi-bin/ToshibaCSG/jsp/SUPPORTSECTION... (i don't know if this page will still visible in future)
- no "mux","mix", or "aux" record source to directly record from desktop
The codec has no function to record from the mixer amp (some other Realtek codecs have, though). It's again a software issue.
ok.. so i've no way to "record what I'm listening" right? (no way other that using a microphone off course)
No, you don't. aplay just shows the available "configuration" but it the device doesn't support surrounds.
oh ok :) do you know how can i test subwoofer? it seems active but if there's a way to actually "test" it would be nice
sorry for me bothering you :) i hope one day I'll be able to pay back your kindness -- Daniele _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
Tobin Davis ha scritto:
The only way to support Midi on that system is via software midi. There is no hardware midi support directly to the system (no place to plug in midi devices), except through USB or possibly integrated in a docking system. To get Midi support for Linux, use Timidity (http://timidity.sourceforge.net). This will convert midi sequences to pcm audio tracks.
ok.. they sucks :) anyway i know how to do software midi.. i've written a guide for it too some times ago, thanks
I personally can't translate the specs listed, but from what I can tell, the system should have a Line-in, Mic-In, and HP-out jacks, along with 4 internal speakers (nothing indicating surround sound). It does have SPDIF out, and I would assume that will allow surround sound if hooked up to a digital receiver, as SPDIF is essentially a digital pass-through. The codec itself only has two DACs, which could be configured between the headphones and speakers to do a quasi-surround4.0 setup, but based on the chip diagram would take some additional programming effort.
i'm 130% sure there's a subwoofer anyway..
And I'm sorry I dropped the ball in helping here. I wish I could spend more time on Alsa, but economics forces me to have a full time job doing non-alsa work at this time. Bad excuse, I know.
i understand it :) no problem! thanks a lot for what you have done!
I'll try to get my test board working again this week, and I'll even download the Windows drivers for your system to use as a baseline for comparison. No guarantees, though.
ok.. if you found something keep me informed, thanks! -- Daniele
On Tue, 2008-02-19 at 19:52 +0100, Daniele (Mastro) wrote:
i'm 130% sure there's a subwoofer anyway..
That's possible, but it would be hooked up to the main speakers the same way most older stereo's had tweeters and subwoofers in the same cabinets with only two wires going to the receiver. The subwoofer just works with the lower frequency mudulation.
participants (3)
-
Daniele (Mastro) -
Takashi Iwai -
Tobin Davis